1. Technical Field
The present invention relates generally to communication networks, and more particularly, to CMDA2000 network that uses Session Initiation Protocol (SIP) for call delivery over a wireless communication network.
2. Related Art
Traditional telephone networks, including the Public Switched Telephone Network (PSTN) and Signaling System Number 7 (SS7) networks, have provided closed systems that enabled users to achieve added capabilities beyond merely connecting a call. Initially, message storage capabilities such as those provided by answering machines and voice mail services (in SS7 networks) were popular. Since then, many other services have been developed as SS7 and other intelligent networks (IN and AIN) have gained widespread popularity. With the advent of Internet telephony, a need to provide voice and video mail services, as well as traditional services, has been realized.
During the past few years, Internet telephony has evolved from being a novelty for the technically oriented seeking party conversation material to a technology that, in the not too distant future, may largely replace the existing telephone networks. Supporting the widespread use of Internet telephony requires a host of standardized protocols to ensure transport audio and video data having a specified quality of service (QoS). These protocols are also needed to provide directory services and to enable signaling. Signaling protocols are of particular interest because they are the basis for advanced services such as mobility, universal numbers, multiparty conferencing, voice mail, and automatic call distribution.
SIP's strengths as such a protocol include its simplicity, scalability, extensibility, and modularity. As a result, increasing interest in SIP is being realized as the SIP standards and protocol requirements develop into maturity. The SIP is defined in RFC 3261. RFC 3261 defines the application-layer control protocol that can be used to establish, modify, and terminate multimedia sessions (e.g., Internet telephone calls)
As a traditional text-based Internet protocol, SIP resembles the hypertext transfer protocol (HTTP) and simple mail transfer protocol (SMTP). SIP uses Session Description Protocol (SDP) for media description. SIP, however, is independent of the transport layer (i.e., IP). Among SIP basic features, the protocol also enables personal mobility by providing the capability to reach a called party at a single, location-independent address.
SIP's basic architecture is client/server in nature. The main entities in SIP are the User Agent, the SIP Proxy Server, the SIP Redirect Server and the Registrar.
The User Agents, or SIP endpoints, function as clients (UACs) when initiating requests and as servers (UASs) when responding to requests. User Agents communicate with other User Agents directly or via an intermediate server. The User Agent also stores and manages call states.
SIP servers have the capability to behave as a proxy or to provide redirection to another SIP server. SIP Proxy Servers forward requests from the User Agent to a next SIP server in a network having a plurality of SIP servers and may also retain information for billing/accounting purposes. SIP Redirect Servers respond to client requests and inform them of the requested server's address. SIP servers can either maintain state information (stateful) or forward requests in a stateless fashion. Generally, SIP is independent of the packet layer and only requires a datagram service. SIP also supports mechanisms for encryption and authentication.
Current CDMA related standards for Legacy Circuit-Switched networks only address using network protocols such as SS7 for call delivery. Thus, a call origination message to the call serving switch (referenced as MSCe that is responsible for the called mobile) is implemented using SS7 defined protocols. This delivery transport has several drawbacks for mobile-to-mobile calls. One that is significant is that mobile-to-mobile calls in the bearer path (path actually carrying the voice between the two mobiles) will require two transcodings by two transcoders. A transcoder is a device that inputs a data stream in one format and output the data stream in another format (e.g., input is a voice data stream encoded using SMV and the output is a data stream encoded using ITU G.711). Other types of encoding for compressed speech that may be used included but are not limited to SMV and EVRC compressed speech encoders. For audio data streams transcoding is undesirable for it decreases voice quality and increases time delay between the two endpoints. A need exists, therefore, for a system and method that minimizes a number of transcodings to minimize call degradation.
Another shortcoming relates to ringback procedures. Heretofore, a serving network element such as the serving MSCe has determined when ringback procedures are to be initiated to generate a ringing sound to the calling party. Because wireless networks are being merged with data packet technologies, however, traditional functionality such as where ringback is provided, may not be optimal according to specific network conditions. A need exists, therefore, for ringback decision making logic that selectively allows another circuit element besides the serving MSCe to generate or initiate ringback for the calling party.